Many communications service providers, such as AT&T, have begun to migrate voice traffic from conventional circuit-switch networks onto packet networks. To transport a voice call over a packet network, an ingress gateway digitizes outbound voice into packets for transport across a packet network, typically an Internet Protocol network, to an egress gateway. The egress gateway converts the packets into voice for ultimate receipt by a telephone set at a called party's premises. Voice originating at the called party's premise is converted by the egress gateway into packets for transport to the ingress gateway for conversion into voice received by the calling party.
Transmission of voice in this manner often referred to as “Voice over Packet” or “VoP” achieves more efficient use of resources. Unlike conventional circuit switched networks that require a dedicated communications path for the duration of a voice call, a single link in a packet network can carry packets associated with different calls. Stated another way, the communications link in a packet network is only dedicated for the interval needed to route packet for a given call for the time needed to route the packet from one hop to another.
Voice over Packet does suffer from a major disadvantage as compared to circuit-switched telephony. With circuit-switched telephony, various techniques exist for monitoring the quality of the call. For example, parameters, such as signal distortion, frequency response, and signal amplitude associated with a telephone call transported over a circuit switched network possess the capability of being monitored for to ascertain voice quality. Monitoring the quality of Voice over Packet calls has proven more problematic. While it is possible to measures to factors such as jitter and delay that affect Voice over Packet transmission, the exact correlation of these parameters to voice quality has heretofore proven difficult to predict with great accuracy.
Recently, the International telecommunications Union promulgated ITU-T Recommendation G.107 that proposed a model (referred to as the “E-Model”) for summing impairment parameters to yield a metric indicative of the voice quality associated with calls over a specified path. Specifically, the E-Model embodied within the ITU-T Rec. 107 defined a measure of voice quality based on a Rating factor, called R, related to the Mean Opinion Score (MOS) of a call. The E-model defined R by the relationship:R=100−Is−Id−Ief+E  (Equation 1)where Is represents the impairment of the voice quality within a path due to signal impairment, Id represents impairment to the voice quality due to delay, Ief represents the impairments due to various network equipment (e.g., codec compression and loss concealment) that impact the signal in a non-linear fashion, and E is a factor to cover lower user expectations of quality due to such things as convenience of service (e.g., wireless) and lower prices. The higher the R factor, the higher the quality of the conversational voice. The E-model gives analytic expressions for the Is and Id components in terms of channel characteristics such as noise and delay but the E-model leaves the Ief factor as a measurable quantity. Thus, the Ief factor depends on the specific network equipment employed to carry the call. Consequently, implementation of the E-Model requires apriori knowledge of the parameters of each piece of network equipment, which can prove especially difficult if packets must travel over different networks.
Thus, there is a need for a technique that enables a fully analytical measurement of the R factor in terms of measurable transport metrics to facilitate ongoing measurements of voice quality without apriori knowledge of the equipment within the network.